/* Copyright (C) 2013 Paul Davis This program is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version. This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more details. You should have received a copy of the GNU General Public License along with this program; if not, write to the Free Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ #ifndef __libardour_audiobackend_h__ #define __libardour_audiobackend_h__ #include #include #include #include #include #include "ardour/types.h" #include "ardour/audioengine.h" #include "ardour/port_engine.h" #include "ardour/visibility.h" #ifdef ARDOURBACKEND_DLL_EXPORTS // defined if we are building the ARDOUR Panners DLLs (instead of using them) #define ARDOURBACKEND_API LIBARDOUR_HELPER_DLL_EXPORT #else #define ARDOURBACKEND_API LIBARDOUR_HELPER_DLL_IMPORT #endif #define ARDOURBACKEND_LOCAL LIBARDOUR_HELPER_DLL_LOCAL namespace ARDOUR { class AudioBackend : public PortEngine { public: AudioBackend (AudioEngine& e) : PortEngine (e), engine (e) {} virtual ~AudioBackend () {} /** Return the name of this backend. * * Should use a well-known, unique term. Expected examples * might include "JACK", "CoreAudio", "ASIO" etc. */ virtual std::string name() const = 0; /** Return true if the callback from the underlying mechanism/API * (CoreAudio, JACK, ASIO etc.) occurs in a thread subject to realtime * constraints. Return false otherwise. */ virtual bool is_realtime () const = 0; /* Discovering devices and parameters */ /** Return true if this backend requires the selection of a "driver" * before any device can be selected. Return false otherwise. * * Intended mainly to differentiate between meta-APIs like JACK * which can still expose different backends (such as ALSA or CoreAudio * or FFADO or netjack) and those like ASIO or CoreAudio which * do not. */ virtual bool requires_driver_selection() const { return false; } /** If the return value of requires_driver_selection() is true, * then this function can return the list of known driver names. * * If the return value of requires_driver_selection() is false, * then this function should not be called. If it is called * its return value is an empty vector of strings. */ virtual std::vector enumerate_drivers() const { return std::vector(); } /** Returns zero if the backend can successfully use @param name as the * driver, non-zero otherwise. * * Should not be used unless the backend returns true from * requires_driver_selection() */ virtual int set_driver (const std::string& /*drivername*/) { return 0; } /** used to list device names along with whether or not they are currently * available. */ struct DeviceStatus { std::string name; bool available; DeviceStatus (const std::string& s, bool avail) : name (s), available (avail) {} }; /** Returns a collection of DeviceStatuses identifying devices discovered * by this backend since the start of the process. * * Any of the names in each DeviceStatus may be used to identify a * device in other calls to the backend, though any of them may become * invalid at any time. */ virtual std::vector enumerate_devices () const = 0; /** Returns a collection of float identifying sample rates that are * potentially usable with the hardware identified by @param device. * Any of these values may be supplied in other calls to this backend * as the desired sample rate to use with the name device, but the * requested sample rate may turn out to be unavailable, or become invalid * at any time. */ virtual std::vector available_sample_rates (const std::string& device) const = 0; /* Returns the default sample rate that will be shown to the user when * configuration options are first presented. If the derived class * needs or wants to override this, it can. It also MUST override this * if there is any chance that an SR of 44.1kHz is not in the list * returned by available_sample_rates() */ virtual float default_sample_rate () const { return 44100.0; } /** Returns a collection of uint32 identifying buffer sizes that are * potentially usable with the hardware identified by @param device. * Any of these values may be supplied in other calls to this backend * as the desired buffer size to use with the name device, but the * requested buffer size may turn out to be unavailable, or become invalid * at any time. */ virtual std::vector available_buffer_sizes (const std::string& device) const = 0; /* Returns the default buffer size that will be shown to the user when * configuration options are first presented. If the derived class * needs or wants to override this, it can. It also MUST override this * if there is any chance that a buffer size of 1024 is not in the list * returned by available_buffer_sizes() */ virtual uint32_t default_buffer_size () const { return 1024; } /** Returns the maximum number of input channels that are potentially * usable with the hardware identified by @param device. Any number from 1 * to the value returned may be supplied in other calls to this backend as * the input channel count to use with the name device, but the requested * count may turn out to be unavailable, or become invalid at any time. */ virtual uint32_t available_input_channel_count (const std::string& device) const = 0; /** Returns the maximum number of output channels that are potentially * usable with the hardware identified by @param device. Any number from 1 * to the value returned may be supplied in other calls to this backend as * the output channel count to use with the name device, but the requested * count may turn out to be unavailable, or become invalid at any time. */ virtual uint32_t available_output_channel_count (const std::string& device) const = 0; /* Return true if the derived class can change the sample rate of the * device in use while the device is already being used. Return false * otherwise. (example: JACK cannot do this as of September 2013) */ virtual bool can_change_sample_rate_when_running () const = 0; /* Return true if the derived class can change the buffer size of the * device in use while the device is already being used. Return false * otherwise. */ virtual bool can_change_buffer_size_when_running () const = 0; /* Set the hardware parameters. * * If called when the current state is stopped or paused, * the changes will not take effect until the state changes to running. * * If called while running, the state will change as fast as the * implementation allows. * * All set_*() methods return zero on success, non-zero otherwise. */ /** Set the name of the device to be used */ virtual int set_device_name (const std::string&) = 0; /** Set the sample rate to be used */ virtual int set_sample_rate (float) = 0; /** Set the buffer size to be used. * * The device is assumed to use a double buffering scheme, so that one * buffer's worth of data can be processed by hardware while software works * on the other buffer. All known suitable audio APIs support this model * (though ALSA allows for alternate numbers of buffers, and CoreAudio * doesn't directly expose the concept). */ virtual int set_buffer_size (uint32_t) = 0; /** Set the preferred underlying hardware sample format * * This does not change the sample format (32 bit float) read and * written to the device via the Port API. */ virtual int set_sample_format (SampleFormat) = 0; /** Set the preferred underlying hardware data layout. * If @param yn is true, then the hardware will interleave * samples for successive channels; otherwise, the hardware will store * samples for a single channel contiguously. * * Setting this does not change the fact that all data streams * to and from Ports are mono (essentially, non-interleaved) */ virtual int set_interleaved (bool yn) = 0; /** Set the number of input channels that should be used */ virtual int set_input_channels (uint32_t) = 0; /** Set the number of output channels that should be used */ virtual int set_output_channels (uint32_t) = 0; /** Set the (additional) input latency that cannot be determined via * the implementation's underlying code (e.g. latency from * external D-A/D-A converters. Units are samples. */ virtual int set_systemic_input_latency (uint32_t) = 0; /** Set the (additional) output latency that cannot be determined via * the implementation's underlying code (e.g. latency from * external D-A/D-A converters. Units are samples. */ virtual int set_systemic_output_latency (uint32_t) = 0; /* Retrieving parameters */ virtual std::string device_name () const = 0; virtual float sample_rate () const = 0; virtual uint32_t buffer_size () const = 0; virtual SampleFormat sample_format () const = 0; virtual bool interleaved () const = 0; virtual uint32_t input_channels () const = 0; virtual uint32_t output_channels () const = 0; virtual uint32_t systemic_input_latency () const = 0; virtual uint32_t systemic_output_latency () const = 0; /** override this if this implementation returns true from * requires_driver_selection() */ virtual std::string driver_name() const { return std::string(); } /** Return the name of a control application for the * selected/in-use device. If no such application exists, * or if no device has been selected or is in-use, * return an empty string. */ virtual std::string control_app_name() const = 0; /** Launch the control app for the currently in-use or * selected device. May do nothing if the control * app is undefined or cannot be launched. */ virtual void launch_control_app () = 0; /* @return a vector of strings that describe the available * MIDI options. * * These can be presented to the user to decide which * MIDI drivers, options etc. can be used. The returned strings * should be thought of as the key to a map of possible * approaches to handling MIDI within the backend. Ensure that * the strings will make sense to the user. */ virtual std::vector enumerate_midi_options () const = 0; /* Request the use of the MIDI option named @param option, which * should be one of the strings returned by enumerate_midi_options() * * @return zero if successful, non-zero otherwise */ virtual int set_midi_option (const std::string& option) = 0; virtual std::string midi_option () const = 0; /* State Control */ /** Start using the device named in the most recent call * to set_device(), with the parameters set by various * the most recent calls to set_sample_rate() etc. etc. * * At some undetermined time after this function is successfully called, * the backend will start calling the ::process_callback() method of * the AudioEngine referenced by @param engine. These calls will * occur in a thread created by and/or under the control of the backend. * * @param for_latency_measurement if true, the device is being started * to carry out latency measurements and the backend should this * take care to return latency numbers that do not reflect * any existing systemic latency settings. * * Return zero if successful, negative values otherwise. * * * * * Why is this non-virtual but ::_start() is virtual ? * Virtual methods with default parameters create possible ambiguity * because a derived class may implement the same method with a different * type or value of default parameter. * * So we make this non-virtual method to avoid possible overrides of * default parameters. See Scott Meyers or other books on C++ to understand * this pattern, or possibly just this: * * http://stackoverflow.com/questions/12139786/good-pratice-default-arguments-for-pure-virtual-method */ int start (bool for_latency_measurement=false) { return _start (for_latency_measurement); } /** Stop using the device currently in use. * * If the function is successfully called, no subsequent calls to the * process_callback() of @param engine will be made after the function * returns, until parameters are reset and start() are called again. * * The backend is considered to be un-configured after a successful * return, and requires calls to set hardware parameters before it can be * start()-ed again. See pause() for a way to avoid this. stop() should * only be used when reconfiguration is required OR when there are no * plans to use the backend in the future with a reconfiguration. * * Return zero if successful, 1 if the device is not in use, negative values on error */ virtual int stop () = 0; /** While remaining connected to the device, and without changing its * configuration, start (or stop) calling the process_callback() of @param engine * without waiting for the device. Once process_callback() has returned, it * will be called again immediately, thus allowing for faster-than-realtime * processing. * * All registered ports remain in existence and all connections remain * unaltered. However, any physical ports should NOT be used by the * process_callback() during freewheeling - the data behaviour is undefined. * * If @param start_stop is true, begin this behaviour; otherwise cease this * behaviour if it currently occuring, and return to calling * process_callback() of @param engine by waiting for the device. * * Return zero on success, non-zero otherwise. */ virtual int freewheel (bool start_stop) = 0; /** return the fraction of the time represented by the current buffer * size that is being used for each buffer process cycle, as a value * from 0.0 to 1.0 * * E.g. if the buffer size represents 5msec and current processing * takes 1msec, the returned value should be 0.2. * * Implementations can feel free to smooth the values returned over * time (e.g. high pass filtering, or its equivalent). */ virtual float dsp_load() const = 0; /* Transport Control (JACK is the only audio API that currently offers the concept of shared transport control) */ /** Attempt to change the transport state to TransportRolling. */ virtual void transport_start () {} /** Attempt to change the transport state to TransportStopped. */ virtual void transport_stop () {} /** return the current transport state */ virtual TransportState transport_state () const { return TransportStopped; } /** Attempt to locate the transport to @param pos */ virtual void transport_locate (framepos_t /*pos*/) {} /** Return the current transport location, in samples measured * from the origin (defined by the transport time master) */ virtual framepos_t transport_frame() const { return 0; } /** If @param yn is true, become the time master for any inter-application transport * timebase, otherwise cease to be the time master for the same. * * Return zero on success, non-zero otherwise * * JACK is the only currently known audio API with the concept of a shared * transport timebase. */ virtual int set_time_master (bool /*yn*/) { return 0; } virtual int usecs_per_cycle () const { return 1000000 * (buffer_size() / sample_rate()); } virtual size_t raw_buffer_size (DataType t) = 0; /* Process time */ /** return the time according to the sample clock in use, measured in * samples since an arbitrary zero time in the past. The value should * increase monotonically and linearly, without interruption from any * source (including CPU frequency scaling). * * It is extremely likely that any implementation will use a DLL, since * this function can be called from any thread, at any time, and must be * able to accurately determine the correct sample time. * * Can be called from any thread. */ virtual pframes_t sample_time () = 0; /** Return the time according to the sample clock in use when the most * recent buffer process cycle began. Can be called from any thread. */ virtual pframes_t sample_time_at_cycle_start () = 0; /** Return the time since the current buffer process cycle started, * in samples, according to the sample clock in use. * * Can ONLY be called from within a process() callback tree (which * implies that it can only be called by a process thread) */ virtual pframes_t samples_since_cycle_start () = 0; /** Return true if it possible to determine the offset in samples of the * first video frame that starts within the current buffer process cycle, * measured from the first sample of the cycle. If returning true, * set @param offset to that offset. * * Eg. if it can be determined that the first video frame within the cycle * starts 28 samples after the first sample of the cycle, then this method * should return true and set @param offset to 28. * * May be impossible to support outside of JACK, which has specific support * (in some cases, hardware support) for this feature. * * Can ONLY be called from within a process() callback tree (which implies * that it can only be called by a process thread) */ virtual bool get_sync_offset (pframes_t& /*offset*/) const { return false; } /** Create a new thread suitable for running part of the buffer process * cycle (i.e. Realtime scheduling, memory allocation, etc. etc are all * correctly setup), with a stack size given in bytes by specified @param * stacksize. The thread will begin executing @param func, and will exit * when that function returns. */ virtual int create_process_thread (boost::function func) = 0; /** Wait for all processing threads to exit. * * Return zero on success, non-zero on failure. */ virtual int join_process_threads () = 0; /** Return true if execution context is in a backend thread */ virtual bool in_process_thread () = 0; /** Return the minimum stack size of audio threads in bytes */ static size_t thread_stack_size () { return 100000; } /** Return number of processing threads */ virtual uint32_t process_thread_count () = 0; virtual void update_latencies () = 0; /** Set @param speed and @param position to the current speed and position * indicated by some transport sync signal. Return whether the current * transport state is pending, or finalized. * * Derived classes only need implement this if they provide some way to * sync to a transport sync signal (e.g. Sony 9 Pin) that is not * handled by Ardour itself (LTC and MTC are both handled by Ardour). * The canonical example is JACK Transport. */ virtual bool speed_and_position (double& speed, framepos_t& position) { speed = 0.0; position = 0; return false; } protected: AudioEngine& engine; virtual int _start (bool for_latency_measurement) = 0; }; struct AudioBackendInfo { const char* name; /** Using arg1 and arg2, initialize this audiobackend. * * Returns zero on success, non-zero otherwise. */ int (*instantiate) (const std::string& arg1, const std::string& arg2); /** Release all resources associated with this audiobackend */ int (*deinstantiate) (void); /** Factory method to create an AudioBackend-derived class. * * Returns a valid shared_ptr to the object if successfull, * or a "null" shared_ptr otherwise. */ boost::shared_ptr (*factory) (AudioEngine&); /** Return true if the underlying mechanism/API has been * configured and does not need (re)configuration in order * to be usable. Return false otherwise. * * Note that this may return true if (re)configuration, even though * not currently required, is still possible. */ bool (*already_configured)(); }; } // namespace #endif /* __libardour_audiobackend_h__ */